There are other part numbers for phones with MGCP and with no software. While the 30x, 50x, and 600 can be converted between SIP and MGCP, but this is unreliable and thus not recommended. In particular, the 50x has different keycaps, which makes this doubly difficult. See below. Also, there are different part numbers for regions other than NA due to power differences.

The IP 500 used to support H.323, but Polycom has discontinued H.323 support on their phones.

The new Polycom SoundStation IP 4000 supports SIP and uses the same SIP software as the SoundPoint IP 30x/50x/60x phones.

New phones:

VVX1500D is a touchscreen videophone that supports SIP and H.323 video. It can inter-operate with other Polycom IP video conferencing units using the H.323 stack and also act as a SIP for voice calls.

SoundPoint IP 670 looks like a color version of the 650 with added gigabit ethernet support. There is also a color expansion module.

SoundStation IP 6000 and 7000 support G.722 wideband speech and PoE. The 6000 looks like an upgrade to the 4000 (just as the 650 was an upgrade to the 601). The 7000 is a larger phone can be linked to a second unit for huge conference tables. Both have better pickup range then before and support additional microphones. Both require SIP 3.0.2 software.

The SoundPoint IP 550 is an upgrade to the 500 line. It includes additional features including: LCD backlight, G.722 wideband codec. Also supports 802.11af PoE.

The SoundPoint IP 320/330 2 line IP phones were recently released. The 330 has a built in 10/100 switch were as the 320 does not. Full-Duplex Speakerphone, 2 lines, IEEE 802.3af. A two-minute screencast on these phones is available from tipandring.org.

The SoundPoint IP 650 is an upgrade to the 600 line. It includes additional features including: LCD backlight, USB port, G.722 wideband codec, metal faceplate accents, 6 additional SIP registrations when used with the expansion module (12 lines total). It still has all the 600 features including POE and 10/100meg ethernet support.

The Polycom SoundPoint IP 430 is a new 2-line desktop speaker phone that fits in the product line above the 300 (no speaker phone) and below the 500 (3 lines). It's about the size of the 300 but has more features and keys like the 500. PoE is supported. See also Polycom 430 Notes

The SoundPoint IP 601 is now released and has been shipping September 2005. It is mostly the same as the 600 but with the addition of the side slide connector with power and IrDA for the expansion console. The expansionattendant console supports 14 additional line keys using SIP software from the 601. It has it's own LCD display with line keys and dual-color LEDs. Power and network are directly connected from the 601 phone on the side connector. Up to three attached expansion modules are supported. The 601 and expansion unit require SIP software 1.6.2. For configuring the expansion module, see SoundPoint IP 601 Expansion Module.

There are now updated models of the SoundPoint IP: 301 and 501. These are the same price as the 300 and 500, but they have more memory to accomodate growing SIP image sizes. The 600 already has this extra memory. So far, the feature differences between the 300/500 and 301/501 are minimal, however the 300/500 may not get some of the "heavier" new features. SSL/TLS (HTTPS and FTPS) will not be supported on the 300/500, for example. Be very sure you are buying from (or become) a certified reseller or you will not be able to get support, software, or documentation for your phones.

The SoundPoint IP phones are very similar to the Cisco 7912, 7940 and 7960, but cost much less from $140 to $290 including power supply. Polycom also includes the software license with the hardware, whereas Cisco requires you to pay extra.

WARNING: The IP 30x and IP 50x models do not have on-board Power Over Ethernet chips. Although the phone claims to support 802.3af and the Cisco POE standard (note it says "optional"), the an additional cable (see part list above) is required on these models. This raises the list price to $215 or $305 when used in a Power over Ethernet environment; if you know you're going to need PoE, buy the part with the PoE cable included (and no wall power brick) to save money. This warning does not apply to the 60x or any future models.

The Polycom phones have a large display and several programmable buttons, and all but the 30x have a very high-quality full-duplex speakerphone. The 30x has a listen-only speaker, which is useful for checking voicemail and listening to boring conference calls.

To set these phones up with Asterisk you need to put configuration files based on the phone's MAC address on an boot server that the phone downloads from. The phone also downloads it's firmware from that same location. The phones can also be manually configured without a boot server but not all features are accessible.

In order to upgrade the boot ROM code for the phones, you simply need to extract the bootrom.ld and bootrom.ver files into the FTP homedir.

Electronic Hook Switch Support

Polycom supports Electronic Hook Switch (EHS) on the SoundPoint IP 320*, 330*, 430, 550, 560, 650, and 670 (*Requires 2.5mm to RJ-9 adapter, available from Polycom). The 601 is listed as being supported in some documents but it seems the support was never added, or was removed. SIP Version 3.0 supports the Jabra and SIP Version 3.1 adds support for Plantronics.

Jabra has released both a headset and Electronic Hook Switch (EHS) adapter (part number 14201-17) that will allow the answering of calls without a handset lifter. Jabra/Polycom Brochure. The EHS functionallity is built into model GN9350 and GN9120 EHS. The EHS mode must be enabled on the phone for the feature to work.

Plantronics offers the APP-5 Headset Hookswitch Control. This adaptor supports several different headsets, including the CSxx series. The Plantronics adapter requires Polycom software SIP Version 3.1. The EHS mode must be enabled on the phone for the feature to work.

On the Jabra headset set the mode to DHSG, for the Plantronics the headset lifter is automatic with the adaptor

Make or take a call using the headset by pushing the button on the headset. No more klunky lifters that fall off!

Functions with the EHS feature working (tested with Plantronics):

A ringing line will alert the headset (beeping in the headset earphone)

Pressing the headset key on the phone will activate the headset also (pickup a line/answer a call)

Pressing the button on the headset will activate the headset button on the phone also (pickup a line/answer a call)

When the phone ends a call the headset will deactivate automaticly

When the phone auto-answers a call the headset will activate automaticly if it's the default

You must use the phone's keys to place a call on hold, resume, transfer, etc

You must use the phone's keys to answer second call when you are already on a line

SIP 3.2 and BootROM 4.2

Release expected in 2009 Q3. New SIP/BootROM software will NOT support the SoundPoint IP 301/501/600/601 and SoundStation IP 4000 (as well as the other older already discontinued phones). See Technical Bulletin 48161

SIP 2.2 and BootROM 4.0 information

New version 2.2 does NOT support the old 300 and 500 phones due to lack of memory, and they are discontinued.The new BootROM 4.0 supports the older phones and allows the config file to list which firmware to load.BootROM 4.0 does not support MGCP application software and must not be used on phones running MGCP.The major new addition with this firmware is support for SRTP voice encryption. If you need to learn about how to configure it though, you must request technical bulletin 25751 from Polycom.

SIP 2.1 information

Main things I saw:1. Added microbrowser support to the SoundStation IP 40002. Added table support to microbrowser3. Added ability to strip or insert leading digits for outgoing calls4. Added ability to disable message waiting indication on a line by line basis5. Increased maximum number of digit map segments to 306. Added microbrowser support to the SoundPoint IP 430 & 501 platform7. Added support for adding phone serial number (Ethernet address) to user agent string in HTTP GET’s used by microbrowser, and modified format of user agent string used during provisioning process and used by microbrowser8. Added microbrowser support for forms within tables

Fixes:1. Phone does not update presence status (e.g. to offline) when reboot initiated2. Phone doesn't ring if one line has Do Not Disturb enabled

SIP 2.0 and BootROM 3.2 firmware information

These are the main things I saw:1. Support for the new SoundPoint IP 6502. Add ability to set Ethernet link mode on IP430 and IP 4000 products.3. Added support for NAT keep-alive4. Added template support in master configuration file5. TCP/TLS Encryption of SIP (SRTP encryption of the audiol is NOT supported despite indications elsewhere)6. Support for a different secondary dialtone (ie. dial 9, hear a new dialtone)

SIP 1.6 and BootROM 3.1 firmware information!!

This isn't a terribly interesting release; it's mostly support for the new 601 phone and a few UI improvements. If you're already on SIP 1.5 and BR 3.0 or BR 2.6, don't bother upgrading unless you already have a bug or want to buy 601s when they come out.

SIP 1.5 and BootROM 3.0 firmware information

BR 3.x supports HTTP, HTTPS, and FTPS boot servers, but once you upgrade to this release you cannot downgrade to versions prior to BR 3.0. If you do not require one of these boot protocols, DO NOT upgrade to BR 3.x and instead stick with BR 2.6.1.

BR 2.6.1 is recommended in all cases for both FTP and TFTP. BR 2.5.0 does not mix well with TFTP, nor is it compatible with SIP 1.5 and later.

There are already a few bugs in 1.5.2, one of which is stutter dialtone not working when you have new voice mail messages. However, there are more fixes and some great new features:

Up to 24 calls per "line key" on a 600 (8 calls on the 300 and 500). The number is configurable.

Multiple line keys can be tied to the same SIP registration

Conference join and split two existing calls on a line (only two calls, not more!)

HTTP and Secure file transfers (HTTPS/FTPS) for 301/501/600/601/4000

sip.cfg and ipmid.cfg config files merged; config files from SIP 1.3 and later are forward-compatible, however

CallerID display problem with earier firmware (displaying incoming call and number only) has been resolved. Phone now displays full CallerID.

Clarification on firmware/bootrom compatibility!!

All current BootROMs and Applications (BootROMs 2.6.2 and 3.1.2, Applications 1.5.3 and 1.6.3) will run on all available SoundPoint IP platforms (30x/50x/60x), as well as the SoundStation IP 4000.

The sip.ld image file actually has different software for each model inside; as of 1.5, what loads on a 300/500 is NOT the same as what loads on a 301/501. The differences are minor, but they will grow over time. That said, SIP 1.5 and BR 2.6 (or even SIP 1.6 and BR 3.0 or 3.1) will RUN on a 300/500, but they will be missing features compared to the 301/501.

Asterisk sip.conf example for Polycom phones:

[138polycom]
type=friend
username=138polycom
password=test
host=dynamic
dtmfmode=inband
defaultip=10.0.0.138
mailbox=138
progressinband=no ;Polycom phones seem to have trouble with the default progressinband=never
;;Using 1.4.1 Firmware, DTMF may stop working if it is set to inband. Change to rfc2833.

If the phones fail to register with Asterisk but can still make outbound calls, you likely need to adjust the digest realm parameter from the default of PolycomSPIP. If this does not solve the problem, please visit:http://www.csh.rit.edu/~adamf/IP500.html

To use "hint" / presence monitoring under Asterisk, "line 1" on the Polycom must be the last extension for the phone listed in sip.conf if you registering multiple lines for the phone. This is because of how Asterisk authenticates sip SUBSCRIBE requests. To monitor activity for an extension you can create a contact in the phone Directory and enable "Watch Buddy". The appropriate "hint" priority for that extension must also be defined in Asterisk's extensions.conf file. Selecting "Buddies" at the main phone menu will then show the current status of the extension(s) you've elected to watch.

To get *8 pickup to work with the above example, you need to add the 'callgroup=' and 'pickupgroup=' to both sections.

******************************

Default Passwords:

To get into the web interface, the default username/password is "Polycom"/"456" (Note that this does not work with Safari 1.2.2.)To get into the Admin interface on the hard phone, the password is "456". (Prior to v1.3.1, the web interfaces to the phone uses "Polycom"/"SpIp" as the username/password.)The "user password" (not used much) defaults to "123".

To reset a lost admin password to default: at phone boot, during the 5 second countdown, push/hold 468* on the phone. Enter the phone's MAC address as the password. This will reset network info but keep peers loaded. Tested on Soundpoint IP 650.

Tested on SoundStation IP 4000To reset a lost admin password to default: at phone boot, during the 5 second countdown, push/hold 68* on the phone. Enter the phone's MAC address as the password (small cap). This will reset admin password but keep peers loaded.

Polycom DHCP settings:

If you decide to use DHCP instead of static IP, make sure to use the latest version of dhcpd and add the following options to your DHCP server:

The tftp-server-name will direct the phones to your TFTP or FTP server andthe time-offset will set your phones to the right time offset against GMT.The settings are in seconds and in my case, the Central Time Zone, I am 6hours west of GMT, so -6 times 3600 = -21,600.Note that the "tftp-server-name" is misleading, and will work fine with FTP, FTPS, HTTP, or HTTPS properly configured. The default FTP username and password are both "PlcmSpIp"; you may have to tweak stuff to have your FTP server "know" about uppercase used in usernames. For security reasons, it's recommended you change the username and password.

NOTE from Bill Butler regarding Software Version 1.5.2.0054:

The time-offset option is extremely important and caused me to tear my hair out for a while. I am actually just using a Linksys RV082 as my DHCP server. I spent 4 hours trying to figure out why the phone would boot with the proper offset (acquired from my sip.cfg file on the server, and then suddenly switch to GMT. Apparently, the polycom was getting it's time info from the DHCP server on my linksys and resetting itself incorrectly to GMT. I solved the problem by manually entering the ip address/gateway/dns into my polycom 501. This forced the phone to adhere to the sip.cfg file and disregard the Linksys DHCP server time zone info. I also had to get the phone to drop it's local settings so it would get with the program. Advanced Settings -> Admin Settings -> Reset to Default -> Reset Local Config

After some more reading it appears that there is a 1.6.x version of the Polycom software which allows one to have the sip.cfg file override the DHCP server NTP announcement. That will offer the best of both worlds and is probably the solution of choice.

Here is a sample config to achieve this:tcpIpApp.sntp.address="10.10.10.111" - Set this to a sntp server that the phone can reach.tcpIpApp.sntp.address.overrideDHCP="1" - This tells the phone to listen to Sip.cfg, not DHCPtcpIpApp.sntp.gmtOffset="-18000" - Eastern Standard Time. (NY)tcpIpApp.sntp.gmtOffset.overrideDHCP="1" - Useful if you have phones in multiple zones.

Polycom FTP discussion:

The following only applies to BootROMs prior to 2.6: Polycom phones can use TFTP or FTP. We recommend the latter, because FTP uses time stamps for upgrades, whereas TFTP will need file name changes. You definitely don't want to deal with file name changes and Polycom strongly recommends against TFTP.

For FTP, put the configuration files and the firmware files in the root directory of the FTP account you use. You can change the user and password provided to the FTP server by choosing setup when the phone first boots up. See the manual at section 2.2.1.2. Some FTP servers can't handled the mixed-case default username.

Polycom Config Files:

bootrom.ld - latest bootrom file. Needs to be in the download directoryalong with bootrom.ver if you want to update your phone. Note: bootrom.ver is not needed if the phone is already running BR 2.6.1 or is not using TFTP.

sip.ld - latest sip firmware image. Needs to be in the download directoryif you want to update your phone with a new SIP firmware.

One step I found necessary was to modify the SNTP tag to point to my time server, as it appears that this configuration overrides any settings aquired from dhcp. — This is a bug in earlier versions of SIP; 1.5 does not have this problem. Also, SIP 1.5 merges ipmid.cfg into sip.cfg.

<mac>.cfg - This file tells the phone what to load. Note that letters MUST be in lower case. If this file does not exist, bootroms after 4.0 will load 000000000000.cfg. It looks likethis:

The phone7001.cfg points to the individual config file for the phone thatmatches the mac address of this file; you can call it whatever you like.The file sip.cfg gives the base configurations for the sip application andipmid.cfg configures everything else about the phone.

Any file listed in the CONFIG FILES section of <mac>.cfg needs to exist. Also any file listed in CONFIG_FILES must be correctly parse as XML. That means it needs a <?xml ?> block and a <sip> or <phone1> block.

I have created a basic set of config files with the defaults changed to better suit asterisk.You can find them on my site at:

There was a previous note that these configs do more harm than good. I have never been contacted by anyone with problems, and I have never had a problem myself. IF you have a problem, please contact me atkris@NOUCEkrisk.org

They have MWI stuff turned on, and some tips taken from Auto Answer and Ring Answer sections on the wiki, Let me know if you have any problems or suggestions.

The file 000000000000-directory.xml is a base contact directory for the phones:

Things to help you adjust set volume(s): re: voice.gain..."rx" speaker volume"tx" mic volume"analog" gain between speaker or mic and the converter chip, change this one"digital" the default value for the volume control on the phone"chassis" speakerphone"handset" handset AND ringer"sidetone" the amount of yourself you hear in the earpeice from the mic, makes phone sound "live"

DND (Do-Not-Disturb)

On FreePBX based systems you can enable DND on Polycom phones by adding the following to your phone config (sip.cfg). This enables server side DND by dialing the feature code *76 and sets the DND notification on the phone itself.

Polycom Phone directory script

Please find enclosed a script to manage your XML phone directories. This shell script allows you to add/delete/check extensions in a group of *-directory.xml files, including customized directories.From now on, you can broadcast any directory change to all Polycom end users.This script is open to changes or enhancements.

The following format is supported:<?xml version="1.0" standalone="yes"?><directory><item_list><item><ln>Last Name</ln><fn>Firstname</fn><ct>extension</ct>......</item>

Problems? Please contact me at bart.coppens@gatewaycomms.com**When creating bitmap images for your phone, the file must be a Windows 4-Bit grey-scale bitmap image8-bit images will not work.**

Caveats

SonicWall and transferHold and transfer functions of at least the IP 600 behind certain versions of Sonic Wall routers does not work. A call placed on hold would drop at exactly 5 seconds. Placing and receiving calls works fine. Replacing the Sonic Wall with a later version solved that problem.

Asterisk and NATSometimes Asterisk is not able to reach Polycom phones, especially when the phone is behind NAT. In that case, one can make calls, but cannot receive calls. To fix this, change the default value of reg.x.server.x.expires from "" to some appropriate value in the file phoneX.cfg. For example: reg.1.server.1.expires="600" forces the phone to update the registration every 10 minutes.

Manual rebootTo reboot phones manually, press and hold the following keys simultaneously until a confirmation tone is heard or for about three seconds:

Note: Holding 4, 6, 8 and * resets all parameters configured on the phone (network and otherwise) and then reboots. It is not just a manual reboot.

Caller IDIt is not possible to use the number presented by caller ID in the form +4612345678 begin_of_the_skype_highlighting +4612345678 end_of_the_skype_highlighting begin_of_the_skype_highlighting +4612345678 end_of_the_skype_highlighting which is stored in the answered and missed call lists to make a new call directly. The Polycom will strip the "+"-sign before sending the signal to the PBX.

Converting from MGCP to SIP:

Polycom does not recommend converting phones from MGCP to SIP or vice versa, but it is possible.When attempting to do it with FTP, we've seen -boot.log files like this --0101001606|cfg |3|00|Removed all log files due to limited space during file update.0101001607|app1 |6|00|Error, not enough space for configuration.

It appears that the upgrade works better/more often when using TFTP instead of FTP.

BR 2.5.0 is required to do this safely; BR 2.6.1 is recommended (above).

Converting a Polycom Soundpoint IP 300 MGCP to SIP

The MGCP to SIP conversion works flawlessly with an IP 300 (at least) using BootROM 3.2.3 Rev B and SIP software 2.1.2 over an FTP (not TFTP) connection. Simply set up those versions of the Polycom software for provisioning, turn on the phone, and you will see a variety of messages on the LCD screen as it formats the phone's memory and installs the new SIP software. When the phone finishes installing and rebooting, it will be a fully-functional SIP phone.

Setting the time / NTP floods:

Our firewall reported flooding of NTP requests; the default configuration files include an SNTP server setting for "clock". If the phone can resolve this via DNS, it will try to get the time from it.

You can use any NTP server to set the time; I've used 0.pool.ntp.org with success. You can not set the time manually.

Resetting the Polycom Phone Password if you forget it!!!

You won't find this information in the manual. I had to contact customer support.

After pressing 4, 6, 8, and * it asks for the ADMIN password. Obviously, if you've lost the admin password you're out of luck. Instead of the admin password, use the MAC address for a full reset!

Setting Australian Call Progress Tones

To set the call progress indications such as dialtone, busy, ringback, or the stutter dialtone you need to override the defaults from the sip.cfg file. I've included the values that are correct for Australia. Following Polycom's best practices for config file management, don't edit the sip.cfg directly, add these lines to an override file (e.g. sip_override.cfg) which you list in the macaddress.cfg file before sip.cfg. You have to override both the chord_sets and the patterns. I used the values in the Asterisk indications.conf file as a reference.

Setting UK Call Progress Tones

To set the call progress indications such as dialtone, busy, ringback, or the stutter dialtone you need to override the defaults from the sip.cfg file. I've included the values that are correct for UK. Following Polycom's best practices for config file management, don't edit the sip.cfg directly, add these lines to an override file (e.g. sip_override.cfg) which you list in the macaddress.cfg file before sip.cfg. You have to override both the chord_sets and the patterns. I used the values in the Asterisk indications.conf file as a reference.